1. Field of the Invention
The invention relates to a digital signal processing apparatus suitable for use in execution of a post filtering process to improve a quality of a decoded audio signal in a digital cellular phone.
2. Description of the Related Art
A VSELP (Vector Sum Excited Linear Prediction) technique has been used as an audio coding system in a digital cellular phone in North America and Japan. According to the VSELP system, an adaptive signal is formed from pitch information and a past exciting signal vector. A noise signal is formed by adding a basic vector. An exciting signal is formed by linearly adding the adaptive signal and the noise signal in accordance with a gain which is set in accordance with information indicative of a sound/soundless state. An audio signal is synthesized from the exciting signal by a short period synthesizing filter. A coding is performed by comparing the synthesized audio signal and an input audio signal, and selecting a code such that an error between them is minimum.
In the VSELP, therefore, a parameter .alpha. of the shortperiod synthesizing filter, an exciting source code I, pitch information L, and gains .beta. and .gamma. are transmitted. Upon decoding, the exciting signal is synthesized from a long period filtering state based on the pitch information L and the past exciting signal, an output of a code book based on the exciting source code I, and the gains .beta. and .gamma.. The exciting signal is supplied to a predictive synthesizing filter of the parameter .alpha. and an audio signal is formed. Further, a post filter is used to improve an auditory impression. An auditory distortion is reduced by adaptively enhancing a pitch periodic component and enhancing a formant component.
That is, FIG. 1 shows a construction of a conventional decoder of the VSELP. In FIG. 1, reference numeral 151 denotes a long period filtering state. The long period filtering state 151 outputs a signal b.sub.L (n) based on a past exciting vector and the pitch information L from an input terminal 161. Reference numeral 152 denotes a code book. The code book 152 outputs a noise signal c(n) on the basis of the exciting source code I from an input terminal 162.
An output of the long period filtering state 151 is supplied to a multiplier 153 for multiplying the gain .beta. from an input terminal 163. An output of the code book 152 is supplied to a multiplier 154 to multiply the gain .gamma. from an input terminal 164. Outputs of the multipliers 153 and 154 are supplied to an adder 155. An exciting signal vector ex(n) is formed by the adder 155. The exciting signal vector is supplied to a short period synthesizing filter 156.
The parameter .alpha. from an input terminal 165 is set into the short period synthesizing filter 156. An audio signal is synthesized by the short period synthesizing filter 156. The audio signal is supplied to a post filter 157. The post filter 157 adaptively enhances the pitch periodic component and enhances the formant component. An output of the post filter 157 is taken out from an output terminal 158.
As mentioned above, according to the coding system like a VSELP, the post filter 157 is inserted upon decoding in order to reduce the auditory distortion. In case of realizing such a post filter 157 by a fixed point arithmetic operation, since a gain fluctuation value of a filtering process cannot be known before the filtering, as for a scaling of the filtering process, it is necessary to preliminarily set a slightly larger margin in consideration of a case where the gain becomes maximum. Therefore, when a signal to be filtered that is inputted to the post filter 157 is small and a gain of the filtering process is not so large, there is a problem such that an enough precision cannot be obtained in the filtering process.
That is, the exciting signal vector ex(n) is a linear sum based on sound/soundless information (.beta., .gamma.) of the signal vector b.sub.L (n) which is formed on the basis of the pitch information L and the past exciting signal vector state and the noise signal c(n) from the code book and is expressed by EQU e(X)=.beta.b.sub.L (n)+.gamma.c(n) (1)
By synthesizing it by the short period synthesizing filter 156, a decoded audio signal s(n) which is inputted to the post filter 157 is derived.
By the above equation (1), the exciting signal vector ex(n) is seen as if it is proportional to the signal vector b.sub.L (n) and noise signal c(n). However, the signal vector b.sub.L (n) and noise signal c(n) mutually exert an influence and are not mutually independent. The exciting signal vector ex(n) is fed back to a long period filtering state r(n) and, as shown in FIG. 2, it is expressed as follows. EQU r(n)=r(n+N) (0.ltoreq.n&lt;L.sub.max -N) EQU r(L.sub.max -N+n)=ex(n)
The long period filter output b.sub.L (n) is obtained as follows from the pitch information L. EQU b.sub.L (n)=r(L.sub.max -L+n) (0.ltoreq.n.ltoreq.N)
where,
N: signal vector length PA1 L.sub.max : past exciting signal vector state b.sub.L (n) is obtained from the signal ex(n). The long period filter output b.sub.L (n) and exciting signal ex(n) are not proportional.
If the gain fluctuation value by the filtering process in the post filter 157 is known before the filtering process, the scaling of the filtering process can be set to an optimum value when operating the post filter 157 by a fixed point arithmetic operation from the gain fluctuation value, and a precision can be improved. Since the gain fluctuation occurs by transmitting the signal through the post filter 157, it is considered to provide a gain control circuit at the post stage of the post filter 157. If the gain fluctuation value of the filtering process is known before the filtering process, by using the gain fluctuation value of the filter, a gain of the gain control circuit at the post stage of the post filter 157 can be optimally set.